Integrating SIP control messaging into existing communication center routing infrastructure

ABSTRACT

A software suite is disclosed for routing communication events over a data-packet-network using an IP session initiation and management protocol. The software suite comprises, a server application running on the network for computing and serving routing determinations per request, a session management application running on the network for initiating and managing routed and established session events, a parsing application running on the network for parsing request data received under session initiation protocol and a conversion application running on the network for converting data received under session initiation protocol into a routing request. All received communication requests for routing are in the form of the session initiation protocol wherein they are parsed and converted into routing requests processed by the server application and routed to determined destinations and wherein events are established as session events conducted under the session initiation and management protocol.

CROSS-REFERENCE TO RELATED DOCUMENTS

The present invention is a Continuation-In-Part (CIP) to a U.S. patentapplication Ser. No. 09/160,558, entitled “Method and Apparatus forProviding Integrated Routing for PSTN and IPNT Calls in a Call Center”,filed on Sep. 24, 1998, disclosure of which is incorporated herein inits entirety by reference. The inventor of the instant application hasalso participated in the document disclosure program and claims priorityto the contents of document disclosure number 496199 dated Jun. 19, 2001reissue patent application of U.S. Pat. No. 7,120,141 filed on Aug. 10,2001 as patent application Ser. No. 09/927,301.

FIELD OF THE INVENTION

The present invention is in the field of telephony communication andpertains more particularly to methods and apparatus for using sessioninitiation protocol (SIP) in the routing infrastructure of acommunication center.

BACKGROUND OF THE INVENTION

In the field of telephony communication, there have been manyimprovements in technology over the years that have contributed to moreefficient use of telephone communication within hosted call-centerenvironments. Most of these improvements involve integrating thetelephones and switching systems in such call centers with computerhardware and software adapted for, among other things, better routing oftelephone calls, faster delivery of telephone calls and associatedinformation, and improved service with regards to client satisfaction.Such computer-enhanced telephony is known in the art ascomputer-telephony integration (CTI).

Generally speaking, CTI implementations of various design and purposeare accomplished both within individual call-centers and, in some cases,at the network level. For example, processors running CTI softwareapplications may be linked to telephone switches, service control points(SCP), and network entry points within a public or private telephonenetwork. At the call-center level, CTI-enhanced processors, dataservers, transaction servers, and the like, are linked to telephoneswitches and, in some cases, to similar CTI hardware at the networklevel, often by a dedicated digital link. CTI and other hardware withina call-center is commonly referred to as customer premises equipment(CPE). It is the CTI processor and application software at such centersthat provides computer enhancement to a call center.

In a CTI-enhanced call center, telephones at agent stations areconnected to a central telephony switching apparatus, such as anautomatic call distributor (ACD) switch or a private branch exchange(PBX). The agent stations may also be equipped with computer terminalssuch as personal computer/video display unit's (PC/VDU's) so that agentsmanning such stations may have access to stored data as well as beinglinked to incoming callers by telephone equipment. Such stations may beinterconnected through the PC/VDUs by a local area network (LAN). One ormore data or transaction servers may also be connected to the LAN thatinterconnects agent stations. The LAN is, in turn, connected to the CTIprocessor, which is connected to the call switching apparatus of thecall center.

When a call arrives at a call center, whether or not the call has beenpre-processed at an SCP, typically at least the telephone number of thecalling line is made available to the receiving switch at the callcenter by the network provider. This service is available by mostnetworks as caller-ID information in one of several formats such asAutomatic Number Identification Service (ANIS). If the call center iscomputer-enhanced (CTI) the phone number of the calling party may beused to access additional information from a customer information system(CIS) database at a server on the network that connects the agentworkstations. In this manner information pertinent to a call may beprovided to an agent, often as a screen pop.

In recent years, advances in computer technology, telephony equipment,and infrastructure have provided many opportunities for improvingtelephone service in publicly-switched and private telephone intelligentnetworks. Similarly, development of a separate information and datanetwork known as the Internet, together with advances in computerhardware and software have led to a new multi-media telephone systemknown in the art by several names. In this new systemology, telephonecalls are simulated by multi-media computer equipment, and data, such asaudio data, is transmitted over data networks as data packets. In thisapplication the broad term used to describe such computer-simulatedtelephony is Data Network Telephony (DNT).

For purposes of nomenclature and definition, the inventors wish todistinguish clearly between what might be called conventional telephony,which is the telephone service enjoyed by nearly all citizens throughlocal telephone companies and several long-distance telephone networkproviders, and what has been described herein as computer-simulatedtelephony or DNT. The conventional system is familiar to nearly all, andis often referred to in the art asconnection-oriented-switched-telephony (COST). The COST designation willbe used extensively herein. The computer-simulated, or DNT systems arefamiliar to those who use and understand computer systems. Perhaps thebest example of DNT is telephone service provided over the Internet,which will be referred to herein as Internet-Protocol-Network-Telephony(IPNT), by far the most extensive, but still a subset of DNT.

Both systems use signals transmitted over network links. In fact,connection to data networks for DNT such as IPNT is typicallyaccomplished over local telephone lines, used to reach such as anInternet Service Provider (ISP). The definitive difference is that COSTtelephony may be considered to be connection-oriented telephony. In theCOST system, calls are placed and connected by a specific dedicatedpath, and the connection path is maintained over the time of the call.Bandwidth is thus assured. Other calls and data do not share a connectedchannel path in a COST system. In a DNT system, on the other hand, thesystem is not dedicated or connection oriented. That is, data, includingaudio data, is prepared, sent, and received as data packets. The datapackets share network links, and may travel by variable paths, beingreassembled into serial order after receipt. Therefore, bandwidth is notguaranteed.

Under ideal operating circumstances a DNT network, such as the Internet,has all of the audio quality of conventional public and privateintelligent telephone-networks, and many advantages accruing from theaspect of direct computer-to-computer linking. However, DNT applicationsmust share the bandwidth available on the network in which they aretraveling. As a result, real-time voice communication may at timessuffer dropout and delay. This is at least partially due to packet lossexperienced during periods of less-than-needed bandwidth which mayprevail under certain conditions such as congestion during peak periodsof use, and so on.

Recent improvements to available technologies associated with thetransmission and reception of data packets during real-time DNTcommunication have enabled companies to successfully add DNT,principally IPNT capabilities, to existing CTI-enhanced call centers.Such improvements, as described herein and known to the inventor,include methods for guaranteeing available bandwidth or quality ofservice (QoS) for a transaction, improved mechanisms for organizing,coding, compressing, and carrying data more efficiently using lessbandwidth, and methods and apparatus for intelligently replacing lostdata by using voice supplementation methods and enhanced bufferingcapabilities.

In typical call centers, DNT is accomplished by Internet connection andIPNT calls. For this reason, IPNT and the Internet will be used almostexclusively in examples to follow. It should be understood, however,that this usage is exemplary, and not limiting.

In systems known to the inventors, incoming IPNT calls are processed androuted within an IPNT-capable call center in much the same way as COSTcalls are routed in a CTI-enhanced center, using similar or identicalrouting rules, waiting queues, and so on, aside from the fact that thereare two separate networks involved. Call centers having both CTI andIPNT capability utilize LAN-connected agent-stations with each stationhaving a telephony-switch-connected headset or phone, and a PCconnected, in most cases via LAN, to the LAN over which IPNT calls maybe routed. Therefore, in most cases, IPNT calls are routed to theagent's PC while conventional telephony calls are routed to the agent'sconventional telephone or headset. However, a method known to theinventor allows one headset to be used at an agent's station forhandling both IPNT and COST calls. This is accomplished via connectingthe agent's telephone to the sound card on the agent's PC/VDU with anI/O cable. In most prior art and current art systems, separate lines andequipment must be implemented for each type of call weather COST orIPNT.

Due in part to added costs associated with additional equipment, lines,and data ports that are needed to add IPNT capability to a CTI-enhancedcall-center, companies are currently experimenting with various forms ofintegration between the older COST system and the newer IPNT system. Forexample, by enhancing data servers, interactive voice response units(IVRs), agent-connecting networks, and so on, with the capability ofunderstanding Internet protocol, data arriving from either network maybe integrated requiring less equipment and lines to facilitateprocessing, storage, and transfer of data. However, telephony, trunksand IPNT network lines representing the separate networks involved stillprovide for significant costs and maintenance.

In some current art implementations, incoming data from the COST networkand the Internet is caused to run side by side from the network level toa call center over a telephone connection (T1/E1) acting as atelephone-data bridge, wherein a certain channels are reserved for COSTconnection, and this portion is dedicated as is necessary in COSTprotocol (connection oriented), and the remainder is used for DNT suchas IPNT calls, and for perhaps other data transmission. Such a serviceis generally offered by a local phone company. This service eliminatesthe requirement for leasing numerous telephony trunks and data-networkconnections. Routing and other equipment, however, must be implementedat both the call-center level and network level significantly reducingany realized cost savings.

A significant disadvantage of such a bridge, having dedicated equipmenton each end, is the dedicated nature of individual channels over thebridging link. Efficient use of bandwidth cannot be assured duringvariable traffic conditions that may prevail at certain times. Forexample, dedicated channels assigned to IPNT traffic would not beutilized if there were not enough traffic to facilitate their use.Similarly, if there was more COST traffic than the allotted number ofCOST channels could carry, no additional channels could be madeavailable.

In a yet more advanced system, known in some call centers, a centralswitch within the call center is enhanced with IP conversion capabilityand can communicate via LAN to connected IP phone-sets and PC'seliminating the need for regular telephone wiring within a call center.However, the service is still delivered via a telephone-data bridge asdescribed above. Therefore, additional requirements for equipment andinefficiency regarding use of bandwidth are still factors.

In still other systems known to the inventor, IPNT to COST conversion orCOST to IPNT conversion is performed within the call center instead ofvia a network bridge. This is accomplished via a gateway connected toboth an IPNT router and a central telephony-switching apparatus. In thefirst case, all calls are converted to and routed as COST calls overinternal telephone wiring to switch-connected headsets. In the secondcase, all COST calls are converted to and routed as IPNT calls over aLAN to individual PC/VDU's.

In all of the described prior art systems, the concerted goal has beento integrate COST and IPNT data via converging at the network level orwithin the call center. The addition of dedicated hardware both at thenetwork level and within the call center adds to the expense ofproviding such integrated data.

In a system known to the inventor and described with reference to Ser.No. 09/160,558 listed in the cross-reference section of thisspecification, an integrated router is provided within a call center.The integrated router monitors and controls both a telephony switchreceiving and forwarding connection-oriented, switched telephony (COST)calls and a Data Network Telephony (DNT) processor receiving andforwarding DNT calls. The integrated router is enabled by software toconsult a common data repository storing status of agents answering bothtypes of calls within the center and routes all calls according to asingle set of routing rules, which can take a variety of forms. In oneaspect, telephone devices at agent stations are adapted to handle bothCOST and DNT calls.

It has occurred to the inventor that in addition to being able to unifyall routed events within a communication center under a common set ofrules, it would be desirable to adapt established IP network protocolsfor use as routing tools within a communication center for the purposeof saving time and costs of developing proprietary protocols andexpensive client applications using them.

One standard Internet-based protocol that may be adapted forcommunication center use is the well-known session initiation protocol(SIP). Very basically, SIP is an application-layer control (signaling)protocol for creating, modifying and terminating communication sessionswith one or more participants. These sessions include Internetmultimedia conferences, Internet telephone calls and multimediadistribution. Members in a session can communicate via multicast or viaa mesh of unicast relations, or a combination of these.

A SIP session can include both persons and automated systems such as amedia storage service. A SIP session can include both unicast andmulticast sessions. A session initiator does not necessarily have to bea member of an initiated session to which SIP is used to initiate. SIPtransparently supports name mapping and redirection services, allowingthe implementation of ISDN and Intelligent Network telephony subscriberservices. These facilities also enable personal mobility.

In the parlance of telecommunications intelligent network services,personal mobility is defined as the ability of end users to originateand receive calls and access subscribed telecommunication services onany terminal in any location, and the ability of the network to identifyend users as they move. Personal mobility is based on uniqueidentification numbering and compliments terminal mobility, whichenables an end terminal to be moved from one sub-net to another.

SIP is designed as part of the well-known IETF multimedia data andcontrol architecture, which is currently incorporating protocols such asRSVP for reserving network resources; the real-time transport protocol(RTP) for transporting real-time data and providing QoS feedback; thereal-time streaming protocol (RTSP) for controlling delivery ofstreaming media; the session announcement protocol (SAP) for advertisingmultimedia sessions via multicast; and the session description protocol(SDP) for describing multimedia sessions.

It is known to the inventors that SIP can be used in conjunction withother call setup and signaling protocols. In this mode, an end systemuses SIP exchanges to determine the appropriate end system address andprotocol from a given address that is protocol-independent. For example,SIP could be used to determine that the party can be reached via H.323,obtain the H.245 gateway and user address and then use H.225.0 toestablish a call, for example. In another example, SIP might be used todetermine that a call recipient is reachable via the PSTN and indicatethe phone number to be called, possibly suggesting an Internet-to-PSTNgateway to be used.

Although SIP protocol is extremely versatile in application, it is yetto be incorporated in call routing infrastructure that depends on avariety of strict call routing rules such as would be the case within acomplex communication center. In a complex central routing system suchas would be established in a state-of-art communication center,practicing IPNT and COST/DNT integration, further innovation is requiredto enable application of SIP as a routing tool that is integrated withestablished routing protocols.

What is clearly needed is a routing system enabled to route both COSTand IPNT calls to available agents sharing a LAN within a call center,wherein SIP protocols are used to set-up, manage, and terminate sessionsbetween agents and clients of the center and between agents and otheragents associated with the center according to established routing rulessetup for the center.

SUMMARY OF THE INVENTION

In a preferred embodiment of the present invention, a software suite isprovided for routing communication events over a data-packet-networkusing an IP session initiation and management protocol. The softwaresuite comprises, a server application running on the network forcomputing and serving routing determinations per request, a sessionmanagement application running on the network for initiating andmanaging routed and established session events, a parsing applicationrunning on the network for parsing request data received under sessioninitiation protocol and a conversion application running on the networkfor converting data received under session initiation protocol into arouting request. All received communication requests for routing are inthe form of the session initiation protocol wherein they are parsed andconverted into routing requests processed by the server application androuted to determined destinations and wherein events are established assession events conducted under the session initiation and managementprotocol.

In a preferred embodiment, the data-packet-network comprises theInternet network. In this preferred embodiment, the Internet networkfurther connects to a LAN network. In one aspect, the software suitecontrols internal routing within a communication center. In anotheraspect, the session management application follows SIP protocols. Instill another aspect, the communication events are sourced from clientsof the center and routed to agents or automated systems at work withinthe center.

In another aspect of the present invention, a method is provided forintelligent routing of communication events from a source to adestination over a data-packet-network using a session initiation andmanagement protocol. The method comprises the steps of, (a) receiving arequest at a routing point for establishing a session event, the requestof the form the session initiation and management protocol, (b) parsingthe request for body content and header information, (c) converting theparsed data into a formal routing request of a form generic to a routingdetermination software, (d) determining the best destination accordingto the request and returning the result to the routing point and (e)establishing the communication event between the source party and thedetermined destination under the session protocol.

In a preferred embodiment, the data-packet-network comprises theInternet network. Also in a preferred embodiment, the Internet networkfurther connects to a LAN network. In one embodiment, the method ispracticed within a communication center. In one aspect of the method instep (a) the routing point is a proxy server and the session initiationand management protocol is SIP protocol. In another aspect of the methodin step (b) the body content of the request is an electronic formpopulated by the requesting party. In one aspect of the method in step(d) additional information pertinent to the requesting party notoriginally part of the request is obtained passed back to the routingpoint along with the determination results. In one aspect of the methodin step (e) the routing point establishes and maintains the sessionuntil a party of the session terminates the session. In another aspectof the method in step (e) the session is established and maintained by anetwork-connected node other than the routing node.

Now, for the first time, a routing system is provided that is able toroute both COST and IPNT calls to available agents sharing a LAN withina call center wherein SIP protocols are used to set-up, manage, andterminate sessions between agents and clients of the center and betweenagents and other agents associated with the center according toestablished routing rules set-up for the center.

BRIEF DESCRIPTION OF THE DRAWING FIGURES

FIG. 1 is a system diagram of a call center connected to atelecommunication network using IPNT to COST conversion according toprior art.

FIG. 2 is a system diagram of the call center and telecommunicationnetwork of FIG. 1 using IPNT switching at the call center according toprior art.

FIG. 3 is a system diagram of the call center and telecommunicationnetwork of FIG. 1 enhanced with integrated routing according to anembodiment of the present invention.

FIG. 4 is an architectural overview of a communication network whereinSIP messaging capability is integrated with routing infrastructureaccording to an embodiment of the present invention.

FIG. 5 a flow diagram illustrating system steps for using SIP in acommunication center according to an embodiment of the presentinvention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 is a system diagram of a call center connected to atelecommunication network using IPNT to COST conversion according toprior art. As described briefly with regards to the background section,various prior art telecommunication networks utilize network-bridgingtechniques for the purpose of causing IPNT and COST incoming calls torun parallel into the call center. In current systems, as was alsodescribed, various implementations have been made within the call centerfor converting IPNT to COST, and conversely, COST to IPNT. FIG. 1represents one such current art system.

In FIG. 1 telecommunications network 11 comprises a publicly-switchedtelephone network (PSTN) 13, the Internet network 15, and a call center17. PSTN network 13 may be a private network rather than a publicnetwork, and Internet 15 may be another public or a private data networkas are known in the art.

In this basic prior art example, call center 17 is equipped to handleboth COST calls and IPNT calls. Both COST calls and IPNT calls aredelivered to call-center 17 by separate network connections. Forexample, a telephony switch 19 in the PSTN may receive incomingtelephone calls and rout them over a COST network trunk 23 to a centralswitching apparatus 27 located within call center 17. IPNT calls fromInternet 15 are routed via a data router 21 over a data-networkconnection 25 to an IPNT router 29 within call center 17. In thisexample, network switch 19 is meant to represent a wide variety ofprocessing and switching equipment in a PSTN, and router 21 is exemplaryof many routers and IP switches in the Internet, as known in the art.

Call center 17 further comprises four agent stations 31, 33, 35, and 37.Each of these agent stations, such as agent station 31, for example,comprises an agent's telephone 47 adapted for COST telephonecommunication and an agent's PC/VDU 39 adapted for IPNT communicationand additional data processing and viewing. Agent's telephones 47, 49,51, and 53 along with agent's PC/VDU 39, 41, 43, and 45 are in similararrangement in agent stations 31, 33, 35, and 37 respectively. Agent'stelephones, such as agent's telephone 49, are connected to COSTswitching apparatus 27 via telephone wiring 56.

A LAN 55 connects agent's PC/VDU's to one another and to a CPE IPNTrouter 29. A client-information-system (CIS) server 57 is connected toLAN 55 and provides additional stored information about callers to eachLAN-connected agent. Router 29 routes incoming IPNT calls to agent'sPC/VDU's that are also LAN connected as previously described. A datanetwork connection 25 connects data router 29 to data router 21 locatedin Internet 15. Specific Internet access and connectivity is not shown,as such is well known in the art, and may be accomplished in any one ofseveral ways. The salient feature to be emphasized in this prior artexample is that separate connections and equipment are necessary andimplemented to be able to handle both COST and IPNT calls at the callcenter.

Each agent's PC/VDU, such as PC/VDU 45 has a connection via LAN 55 anddata network connection 25 to Internet 15 while the assigned agent islogged on to the system, however, this is not specifically required butrather preferred, so that incoming IPNT calls may be routed efficiently.Dial-up connecting rather than a continuous connection to Internet 15may sometimes be employed.

An agent operating at an agent station such as agent station 33 may haveCOST calls arriving on agent's telephone 49 while IPNT calls arearriving on agent's PC/VDU 41. In examples prior to this example, router29 would not have a connection to central switching apparatus 27. Havingno such connection creates a cumbersome situation, requiring agents todistribute their time as best they can between the two types of calls.Thus, agent time is not utilized to maximum efficiency with respect tothe total incoming calls possible from both networks.

In this embodiment however, router 29 is connected to an IPNT-to-COSTgateway 59 via data connection 61. Gateway 59 is connected to centralswitch 27 via CTI connection 63. Gateway 59 is adapted to convert allincoming and outgoing IPNT calls to COST calls where they may be routedover wiring 56 to agents (incoming), or over trunk 23 to switch 19 incloud 13 (outgoing). In this way, agents may use switch-connectedtelephones, such as telephone 47 to answer both IPNT-to-COST convertsand regular incoming COST calls. The agent's time is better utilized,and additional network equipment comprising a network bridge andassociated network connections are not required.

This prior art example, however, presents some problems and limitations.One problem is that traditional COST equipment such as routers,switches, and wiring may have to be significantly expanded to handlemore traffic regarding the added call-load received from cloud 15.Further, the ability to predict possible call overload situations issignificantly complicated because of the convergence of IPNT calls intothe COST routing system. As IPNT calls are now received by agents asCOST calls, certain features inherent to IPNT applications will be lostsuch as multimedia enhancements, and the like.

One advantage with this example is that calls originating as IPNT callswithin call center 17 may be sent as IPNT calls over data connection 25,or as converted COST calls over trunk 23. Another advantage is that LAN55 is free to carry data other than IPNT audio packets.

FIG. 2 is a system diagram of the call center and telecommunicationnetwork of FIG. 1 using IPNT switching at the call center according toprior art. This prior art example is essentially reversed from the priorart example described in FIG. 1. For the sake of saving space andavoiding redundancy, elements found in this example that are identicalto the example of FIG. 1 will not be re-introduced.

Call center 17 receives COST calls from cloud 13 over trunk 23, and IPNTcalls from cloud 15 over data connection 25 as described with the priorart example of FIG. 1. However, instead of having a centraltelephony-switch such as switch 27 of FIG. 1, a COST-to-IPNT gateway 71is provided and adapted to convert COST calls to IPNT calls.

After converting incoming COST calls to IPNT calls, these are routed viadata connection 73 to an IPNT switch 75. IPNT switch 75 is adapted todistribute the resulting IPNT calls to selected agent's over LAN 55.Regular IPNT calls are routed to LAN-connected agents via router 29.

Agent's telephones 47-53 are, in this example, adapted as IP phones andare each connected to LAN 55. Internal wiring and other COST relatedarchitecture is not required, which is one distinct advantage of thisprior art system.

A disadvantage of this system is that there is no provision to makeoutbound calls to the PSTN 13. Only further enhancement to gateway 71 toconvert IPNT calls to COST calls enables out-bound dialing to PSTN 13from within call center 17. Under heavy call-load situations, a dualgateway such as would be the case with gateway 71 may become congestedand cause delay. Additional apparatus may be required to alleviate thisproblem. In some cases wherein there are concerted outbound campaignstaking place on a frequent basis, it may be more prudent to maintain aCOST switch and internal wiring within call center 17 connected toeither agent telephones (maintaining dual capability) or, to add asecond set of telephones dedicated for outbound campaigns. Moreover,agents are reintroduced with a problem solved in the example of FIG. 1of having to deal with incoming calls to both IP phones, and PC/VDU's.

FIG. 3 is a system diagram of the call center and telecommunicationnetwork of FIG. 1 enhanced with integrated routing according to anembodiment of the present invention. As discussed with reference to FIG.2, common elements introduced with the prior art example of FIG. 1 willnot be reintroduced here unless they are altered according to anembodiment of the present invention.

According to a preferred embodiment of the present invention, callcenter 17 receives COST and IPNT calls from their respective separatenetworks comprising telecommunication system 11. Call center 17 is, inthis example, enhanced with an integrated router (IR) 83 capable ofrouting both COST calls and IPNT calls. Central switch 27 is connectedvia CTI link to a processor running instances of a CTI application knownto the inventors as T-server and Stat-server (TS/STAT). An intelligentperipheral in the form of an IVR 84 is connected to processor 82 viadata link 81. Processors 82 and IVR 84 provide CTI enhancement to switch27, as well as an application programming interface (API) to IR 83 viainstalled software.

It will be apparent to the skilled artisan that processor 82, IVR 84 andIR 83 may be implemented in a single computing machine executing all ofthe necessary software, but the functions have separated here forclarity in description.

A multimedia data server (MIS) 87 is connected to LAN 55, and is adaptedto store and serve certain multimedia content as known in the art.Switch 27 and Router 29 are maintained as call-arrival points for callsarriving from either PSTN 13 or Internet 15 adhering to the separatenetwork-architecture previously described.

IR 83 performs in an innovative manner in that it not only controlscentral switch 27 through interaction with processor 82, and thereforerouting of COST calls, but also controls processor 29 and the routing ofIPNT calls. IR 83 controls routing of both COST and IPNT calls whethersuch calls are incoming or outgoing.

An agent status-table 86 is a real-time database containing agentavailability information, which is continually updated as operation ofthe call center proceeds. Table 86 may reside in IR 83 as shown, or mayreside on processor 82 as part of the T-Server software. Table 86 keepstrack of when agents log on or off to the system, and which agents arebusy on calls (either COST or IPNT). It will be appreciated that anycombination of rules set by the company hosting center 17 may be inplace such as priority routing, routing based on skill, statisticalrouting, and so on, in various combinations known to the inventors.

Integrated routing as provided by IR 83 allows calls of both types(COST/IPNT) to be distributed evenly among available agents withoutadding expensive call conversion equipment, or effecting outbounddialing capabilities.

Yet another improvement in this example over prior art systems is knownto the inventor and implemented at some or all agent stations such asstations 31-37. As briefly described with reference to the backgroundsection, agent stations 31-37 have PC-connected telephones. An I/O cablecompletes this interface via connection from a telephonereceiver/transceiver apparatus such as on telephone 53 to a sound cardinstalled on an associated PC such as PC/VDU 45. Individual one's ofheadsets such as headsets a-d are connected either to each telephone oreach PC/VDU and are adapted to allow an agent to engage both COST andIPNT calls using the same headset.

It will be apparent to one with skill in the art that the integratedrouting system of the present invention may be utilized in any callcenter capable of receiving both COST and IPNT (or other DNT)communication. It will also be apparent to one with skill in the artthat the present invention may implemented as part of a CTI softwarepackage, or held separately and integrated with such a CTIimplementation.

SIP-Based Call Control Management

In another aspect of the present invention, the inventor provides amechanism for incorporating SIP protocol as a call management toolwithin a communication center. The methods and apparatus of theinvention are described in enabling detail below.

FIG. 4 is an architectural overview of a communication network 401wherein SIP messaging capability is integrated with routinginfrastructure according to an embodiment of the present invention.Network 401 comprises a PSTN 414, a data -packet-network 417, which inthis example is the well-known Internet network, and atelecommunications center 402.

PSTN 414 can be another type of COST telephone network as may be knownin the art such as a private telephone network. A local telephony switch(LSW) 415 is provided within PSTN 414 and adapted as a switch that islocal to communication center 402. Switch 415 may be an ACD type or PBXtype telephony switch as well as other known types. It will beappreciated by the skilled artisan that there will be many otherswitches, service control points, and other telephony equipmentconnected within PSTN 414. In this simple example, only switch 415 isillustrated and deemed sufficient for the purpose of describing thepresent invention.

CTI equipment (not shown) such as a CTI processor including IVRcapability and a Stat-Server may be assumed to be present within PSTN414 and connected to LSW 415 in cases of network-level routing. In sucha case, a separate network would exist from the described equipment inthe PSTN to similar equipment implemented within center 402.

Internet network 417 comprises an Internet backbone 416 extendingtherethrough and a backbone-connected Internet server 418 that isadapted, in this case, as an Internet access point for IPNT callersattempting to reach communication center 402. Server 418 is adapted toserve HTML electronic documents or electronic documents presented inother mark-up languages, some of which depend on protocols used byconnecting end devices. WML, HDML, and other well-known protocols areexemplary of several that may be employed at server 418. Backbone 416represents all of the lines, equipment and connection points making upthe Internet network as a whole. Therefore, there are no geographiclimitations to the practice of the present invention.

Backbone 416 is illustrated, in this example, as extending toward PSTN414. In some embodiments, calls may travel back and forth between PSTN414 and Internet 417 through a bridge or gateway (not illustrated inthis example). Internet server 418 is adapted as a customer access pointto communication center 402 as previously described. A user representedherein by a PC icon labeled 419 is illustrated in this example asconnected to Internet backbone 416 by an Internet access path 422.Therefore user 419 has accessibility when connected to Internet server418 for the purpose of establishing communication with communicationcenter 402 over backbone 416.

User 419 may establish Internet access with Internet server 418 using avariety of well-known Internet access methods. Typically, user 419 wouldaccess server 418 using a dial-up modem technology through an Internetservice provider (ISP) as is most common in the art. In otherembodiments, user 419 may access via a cable modem connection, awireless satellite connection, an integrated service digital network(ISDN), and so on. Although an ISP is not explicitly illustrated in thisexample, one such may be assumed to be present and operable between user419 and network 417 as is well known in the art. Actual access wouldtake place through network 414 in the case of dial-up services.

Communication center 402 represents a state-of-art center capable ofintegrating COST events with DNT events under a common set of routingrules. A central telephony switch (SW) 413 is provided withincommunication center 402 and adapted as a central office switch forrouting COST communication events within the communication center, andin some cases to remote agents. SW 413 is connected to LSW 415 withinPSTN 414 by at least one telephony trunk 23. Switch 413 may be an ACD orPBX type switch as well as other known types as was described furtherabove.

Communication center 402 has a LAN 403 provided therein and adapted forTCP/IP and other applicable Internet protocols. LAN 403 is chiefly usedin this example to provided network capability for connected agents,automated systems, and other equipment that is further described below.

In this example, there are two illustrated workstations A (404) and N(405) within center 402 that are connected to LAN 403 for networkcommunication. It will be appreciated that there will typically be manymore than 2 workstations connected to LAN 403 as noted by the A-Ndesignation, in a communication center. Each workstation A-N is at leastadapted with a PC and a telephone in this embodiment. In workstation 404there is illustrated a PC 406 connected to LAN 403 and a PC-connected IPphone 407. In workstation 405 there is illustrated a LAN-connected PC408 and a connected IP phone 409. There may be more equipment typesprovided in and operational in a workstation that are not illustrated inthis embodiment including facsimile stations and so on. The inventordeems illustration of two main communication appliances, namely a PC anda telephone, as sufficient for the purpose of explaining the presentinvention.

It is noted herein that there are no COST wiring facilities implementedfrom switch 413 to phones 407 and 409. In this example both phones 409and 407 are IP-capable telephones that are connected to their respectivePCs 409 and 407. The connection is through the PC sound card enablingthe IP phones to take calls through the PC. In this case, all COSTcommunication events at switch 413 are converted to IPNT events androuted to LAN-connected PCs.

A transaction server (T-Server) 412 is provided within communicationcenter 402 and connected to switch 413 by a CTI link. T-Server 412 isalso illustrated herein as LAN connected. T-Server 412 embodies andserves upon request all of the routing functions employed at center 402.A data server 423 is provided within center 402 and connected to LAN403. Server 423 serves any pertinent data regarding client and agentinformation as may be required to enhance routing function. A datarepository 424 is provided and accessible to server 423. Repository 424is adapted to hold any pertinent data that may be accessed and served byserver 423 upon request. Updates to such data may be made periodicallythrough LAN 403.

Types of data stored in repository 424 and served by server 423 mayinclude, but is not limited to, agent information such as log-in status,availability data, skill data, language data, identification data,address data, and so on. Client information contained in repository 424and servable by server 423 may include client history data, clientidentification data, contact information, payment history data, orderstatus data, and so on. Server 423 functions, in this example, as acentralized information source for agents as well as for automatedsystems at work in the center. Information contained in repository 424may be continually updated as events arrive and are internally routedwithin center 402.

A proxy server 410 is provided within center 402 and illustrated asconnected to LAN 403. Proxy server 410 is adapted with a modifiedversion of session initiation protocol (SIP) as is illustrated in thisexample by a software instance (SW) 411. SW 411 is installed on andexecutable on server 423 in accordance with events for internal routingwithin the center. Server 410 has an Internet connection to Internetbackbone 416 by an Internet access pipeline 425. Server 410 functionsalso as an Internet router (IR) as described further above withreference to IR 83 of FIG. 3.

As an IR, server 410 performs all of the internal routing of eventsarriving thereto from Internet 417 and from PSTN 414 through switch 413.To this effect, server 410 is directly connected by a CTI link to switch413. In one embodiment, switch 413 is adapted to convert COST events toIPNT ring events. In another embodiment, server 410 simply routes eventsfrom switch 413 but connection for such events is physically made onconventional telephones and internal telephony wiring. In still anotherembodiment, switch 413, if adapted as an IP conversion switch, may bedirectly connected to LAN 403. There are many possibilities.

User 419 has an instance of a software compatible with SIP protocol (SW)420 executable thereon that is adapted as a simple client application toSW 411 in server 410. SW 420 may be a browser plug-in in one embodiment,for example. In another embodiment, SW 420 may be a stand-aloneapplication. Another instance of software labeled SW 421, is illustratedon PC (user) 419 and adapted as a form-filler (FF) application. FF 421may be assumed to be part of SW 420 as one application in manyembodiments, or be connected to it in a direct or indirect manner. Theinventor logically separates FF 421 from SW 420 for illustration offunction only. In another embodiment, the functions of SW 420 and FF 421may be provided in and accessible from server 418 within Internet 417.

The purpose of FF 421 is to enable a user, in this case user 419, tocommunicate a text reason for a desired connection event to an agent orsystem, of communication center 402. FF 421 provides functionality thatwould otherwise be covered by an interactive voice response (IVR) systemthat may be assumed to be implemented either in PSTN 414 and connectedto switch 415, or within center 402 connected to switch 413.

User 419 may access server 418 and then be provided with applicableclient software or he or she may already have the appropriate softwareinstalled as a resident program. Filling out an electronic form using FF421 and submitting the form while connected online with server 418causes a telephony event request to be initiated having an SIP headerand the completed form as the body of the SIP message. The SIP eventarrives at server 410 where SW 411 parses the message for content andseparates the header information and content (form data) from the SIPmessage.

The parsed data is then re-formatted into language that is understood byT-server 412 and sent as a routing request to the server. Record of theevent remains at server 410 until a response is received from T-server412 concerning routing determination. T-server 412 executes anyapplicable routing routines using the re-formatted SIP data and sends arouting result or recommendation back to server 410. In some embodimentsT-server 412 consults with server 423 for any information required foroptimizing a best determination for routing the particular event.

Server 410 receives a routing determination from sever 412, and thenroutes the target event to an available agent or system based on theresponse. All SIP functionality built into SW 411 can be leveraged toprovide information that is useful for establishing a successfulconnection.

For events arriving at switch 413 wherein there is no agent-levelrouting performed at PSTN 414 network level, IVR interaction can providethe equivalent of FF 421 of PC 419. SW 411 is capable of parsing atextualized of digitized version of an IVR message and of generating anSIP message containing the information. As described above, T-server 412receives a routing request from server 410 in the form of a SIP message.Server 412 computes routing results according to included informationand sends the results to server 410. Server 410 then routes the event toan appropriate agent or system connected to LAN 403.

If events arriving at switch 413 are to be passed directly to LAN 403through a dedicated LAN connection (not shown), then server 410 simplyroutes notifications of pending ringing events. Alternatively, server410 may receive the actual events digitized and my directly route themto appropriate agents or systems over LAN 403. Again, all of thefunctionality of SIP messaging may be tapped wherein it may be useful asa routing variable. Such functions include bandwidth reservation,handshake protocols, media designations, callback information, presenceinformation and so on.

The method and apparatus of the present invention allows integration ofstrict routing conventions and SIP functionality without requiringsignificant modification of or provision of special application programinterfaces (APIs) to be distributed to key components of the system,namely T-server 412, server 423, and perhaps at switch 413.

One with skill in the art will recognize that there may be a variety ofrouting infrastructures having differing hardware components andconnectivity that can be enhanced with SIP-Routing capability accordingto embodiments of the present invention. Likewise, the preferred methodmay be employed to directly route and forward actual events and forrouting notification of pending events wherein subsequent callconnection is a COST connection made between a terminal and a centralswitch of the center.

FIG. 5 shows a simplified flow diagram illustrating system steps forusing SIP in a communication center according to an embodiment of thepresent invention. At step 501, a client of a communication center sendsan SIP request to an SIP proxy analogous to server 410 of FIG. 4. Thisstep is assumed in the case of the request originating in the Internetor other data-packet network. At step 502 the request of step 501 isreceived and parsed for content. This process involves separating thecontent data from the traditional SIP header data. Also at this step theproxy server, after parsing the data, reformats the information into arouting request expressed in the format understood by a transactionserver responsible for executing intelligent routing routines accordingto existing routing rules. After reformatting the data, the proxy atstep 502 sends the reformatted request to the T-server analogous toserver 412 of FIG. 4. At step 503 the T-server receives the routingrequest of step 502 and begins processing the request.

In the meantime, at step 504 the proxy server waits for theresult/response from the request sent at step 502. In step 504 therequesting party or originator of the event remains in queue. At step503 the T-server uses additional information provided by form filling tohelp granulate a routing determination to more narrowly define anappropriate routing destination for the event. This may involve accessand consultation with a server/database analogous to server 423connected to repository 424 described with reference to FIG. 4. At thistime, repository 424 may also be updated with new data from informationprovided with the original SIP request. At step 506 the T-serverretrieves any required additional information from a database ofinformation analogous to the repository/server combination describedabove. This data may be passed to an appropriate agent with or ahead ofthe routed event.

At step 507, the T-server responds to the request of 502, afterprocessing and retrieving any additional data at step 506, by sendingthe best possible routing information or result to the proxy server. Theresult may well be a final routing determination or commandnecessitating no further determination by the proxy. In anotherembodiment, routing information may simply consist of a data recordindicating all of the parameters of the route computation wherein somefurther computation to determine final destination is left for the proxyserver.

At step 508, the proxy sever of step 507 routes any additional hard datato the intended recipient of the call in the form of a screen pop-up orother well-known convention. Simultaneously at step 505, the processedevent is routed by the proxy server to the same recipient. The recipientis most likely a live agent but may also be an automated robotic system.

In one embodiment, the live connection is established and the sessionmaintained within the proxy. In another embodiment only notification ofan event is routed and actual physical connection made by another IProuter (dumb switch) other than the proxy server. In the event oftelephony events arriving through the COAST network (PSTN), the SIPrequest sent to the proxy is generated at an enhanced central switchwherein the IVR interaction, if any, is translated into the form contentof the SIP message. Therefore, the client in 501 in the case of COSTevents would be the central switch analogous to switch 413 describedwith reference to FIG. 4. The active SIP session whether COST initiatedor IPNT initiated is maintained in the proxy server or anotherdesignated server.

Using SIP data to manage internal routing enables all IP communicationforms such as IP telephony, Chat, multiparty conferencing and so on tobe routed and maintained as traditional telephony call events followingstrict intelligent routing regimens. In the case of multipartyconferencing, many steps otherwise required for conferencing in variousparties is eliminated. Each selected party would receive an identicalrouted event, which when taken or picked-up automatically initiates theparty into the conference. Similarly, other traditional steps associatedwith center telephony such as call holding, call waiting, call transfer,etc. can be simplified using SIP parameters. Many individualcharacteristics of SIP capability can be leveraged for mediaidentification, reserving bandwidth, end user identification, protocolswitching to improve transmission quality, and so on.

The method and apparatus of the present invention can be practicedinternally within a communication center and externally betweencommunications centers connected to a common network. The invention mayalso be practiced on virtual IPNT communication networks utilizingremote agents. All that is required in the case of a virtual center is acentralized routing point (proxy server) and the transaction servercapabilities and routines required to provide intelligent routing amongremotely connected agents.

The method and apparatus of the present invention should, in light ofthe many applicable embodiments, be afforded the broadest scope underexamination. The method and apparatus of the present invention should belimited only by the claims that follow.

What is claimed is:
 1. A system for routing a communication event in acall center having routing provided by a computer-telephony intergration(CTI) server, the event initiated by an originator at a computerizedworkstation outside the call center, the system comprising: asoftware-enabled session initiation protocol (SIP) mechanism operable onthe workstation by the originator to prepare and send anmeans in theworkstation for preparing and sending a SIP-protocol routing requestalong with an event initiation; and a software enabled reformattingmechanism in the call centermeans in the call center for receiving andprocessing the SIP-protocol routing request; characterized in that thereformatting mechanism converts, converting the SIP routing request intonon-SIP protocol understood by the CTI server, and sends sending theresulting non-SIP request to the CTI-server CTI server for processingand response, and wherein the CTI server determines and returns arouting for the communication event.
 2. The system of claim 1 whereinthe communication event arrives at the call center from a data packetnetwork.
 3. The system of claim 2 wherein the data-packet-networkcomprises the Internet network.
 4. The system of claim 3 wherein theInternet network further connects to a loca area network (LAN) network.5. The system of claim 1 wherein the CTI server controls routing withinthe call center.
 6. The system of claim 1 wherein the communicationevents are event is received from clients a client of the call centerand routed to agents an agent or automated systems system at work withinthe call center.
 7. A method for routing a communication event in a callcenter having routing provided by a computer-telephony intergrationintegration (CTI) server, the event initiated by an originator at acomputerized workstation outside the call center, the method comprisingthe steps of: a) step for preparing and sending a session initiationprotocol (SIP) routing request along with the initiated event by asoftware-enabled SIP mechanism operable on the workstation by theoriginator; b) step for receiving end and processing the SIP-protocolrouting request by a software enabled reformatting mechanism in the callcenter; c) step for converting the SIP routing request into non-SWnon-SIP protocol understood by the CTI server by the reformattingmechanism; d) sending the non-SIP request to the CTI-server forprocessing and response; and e) determining a routing for thecommunication event by the CTI-server CTI server.
 8. The method of claim7 wherein the communication event arrives at the call center from a datapacket network.
 9. The method of claim 8 wherein the data packet networkcomprises the Internet network.
 10. The method of claim 9, wherein theInternet network further connects to a local area network (LAN) network.11. The method of claim 7 wherein the (CTI) CTI server controls routingwithin the call center.
 12. The method of claim 7 wherein thecommunication events are event is received from clients a client of thecall center and routed to agents or automated systems an agent orautomated system at work within the center.
 13. A system for routing acommunication event in a call center having routing provided by acomputer-telephony integration (CTI) server, the event initiated by anoriginator at a computerized workstation outside the call center, thesystem comprising: a workstation accessible to the originator, theworkstation including a processor and memory, wherein the memory storesinstructions that, when executed by the processor, cause the processorto: prepare and send a SIP-protocol routing request along with an eventinitiation; receive and process the SIP-protocol routing request;convert the SIP routing request into non-SIP protocol understood by theCTI server; and send the resulting non-SIP request to the CTI server forprocessing and response, wherein the CTI server determines and returns arouting for the communication event.
 14. The system of claim 13 whereinthe communication event arrives at the call center from a data packetnetwork.
 15. The system of claim 13 wherein the CTI server controlsrouting within the call center.
 16. The system of claim 13 wherein thecommunication event is received from a client of the call center androuted to an agent or automated system at work within the call center.17. A method for routing a communication event in a call center havingrouting provided by a computer-telephony integration (CTI) server, theevent initiated by an originator at a computerized workstation outsidethe call center, the computerized workstation preparing and sending asession initiation protocol (SIP) routing request along with theinitiated event, the preparing and sending including detecting a datatransmission request from a user interface of the computerizedworkstation, the method comprising: receiving and processing, by anelectronic processing device in the call center, the SIP-protocolrouting request, the receiving and processing including parsing therequest for separating header information from content data; converting,by the electronic processing device, the SIP-protocol routing requestinto non-SIP protocol understood by the CTI server, wherein theconverting includes reformatting the SIP-protocol routing request into anon-SIP request; sending, by the electronic processing device, thenon-SIP request to the CTI server for processing and response; anddetermining, by the electronic processing device, a routing for thecommunication event by the CTI server.
 18. The method of claim 17wherein the communication event arrives at the call center from a datapacket network.
 19. The method of claim 17 wherein the CTI servercontrols routing within the call center.
 20. The method of claim 17wherein the communication event is received from a client of the callcenter and routed to an agent or automated system at work within thecall center.